1. Field of the Invention
The invention relates to a method of transmitting a signal.
2. Background Information
During the transmission of an audio signal, for example for radio broadcasts, cable transmissions, satellite transmissions and for recording devices, it is known to convert the analog audio signal into a digital audio signal of a certain resolution, to transmit it in that form and to convert it back into an analog signal for playback. The digital transmission results in a better signal to noise ratio particularly for playback.
The bandwidth required for the transmission of such a signal is essentially determined by the number of sampling values to be transmitted per unit time and by the resolution.
In practice, a requirement exists to keep the bandwidth necessary for transmission as small as possible in order to be able to make do with a narrow-band channel or transmit as many audio signals as possible simultaneously over a broad-band channel. The required bandwidth per se can be reduced by reducing the number of sampling values or the number of bits per sampling value. However, generally this measure results in worsening of playback quality.
In a method disclosed in DE-OS 3,506,912, the playback quality is improved in that the digital audio signal is divided into sections that are consecutive in time and the signal is transformed into a short-term spectrum which represents the spectral components of the signal for the respective time sections. In the short-term spectrum it is easier than in the time domain to locate components not discerned by the listener, that is, irrelevant components in a communications technology sense on the basis of psychoacoustic rules. These components are given less weight for transmission or are omitted entirely. Thus a considerable portion of the otherwise required data can be eliminated for transmission so that the average bit rate can be reduced considerably.
For the formation of the time sections, the signal is initially evaluated in the time domain by means of an analysis window and after transformation, coding, transmission, decoding and retransformation, it is ultimately evaluated by means of a synthesis window. The configuration of the analysis window influences frequency resolution as well as the quantity of data for transmission. Thus, windows having "hard" edges as exhibited, for example, by a rectangular window, have a poor frequency resolution. This is so because in the evaluated section the spectral components generated by the extremely steep signal rise and drop at the beginning and end of the window are added to the spectrum of the original signal. However, the time sections could be joined with one another without overlaps.
In the method disclosed in DE-OS 3,506,912 a window function has already been selected which has "softer" edges. Here, the beginning and end of the analysis window follow a cosine square function and the corresponding regions of the synthesis window follow a sine square function. The middle region of both windows has a constant value. With such a configuration of the window function, there already results an improved frequency resolution. However, in the region of the "soft" edges it is necessary for successive time sections to overlap. Because of the double transmission of the signals contained in this region, this leads to an increase in the average bit rate.
A further improvement in frequency resolution could be realized by an even less steep edge in the window function of the analysis window and by an expansion of the edge region within the window. However, such a measure inevitably requires a greater overlap of adjacent time sections.
If the edge region is expanded to the point that the window function no longer has a region with a constant value, adjacent time sections must overlap by 50%. This doubles the number of sampling values and correspondingly the quantity of data.
From the publications by J. P. Princen and A. B. Bradley, entitled "Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation", in IEEE Transactions, ASSP-34, No. 5, October, 1986, pages 1153-1161, and by J. P. Princen, A. W. Johnson and A. B. Bradley, entitled "Suband[sic]/Transform Coding Using Filter Bank Design Based on Time Domain Aliasing Cancellation", in IEEE Int. Conference on Acoustics, Speech and Signal Processing, 1987, pages 2161-2164, it is known, in connection with a 50% overlap of successive time sections, to reduce the data quantity back to the original value in that only every second sampling value is coded. This proposal is based on the window functions for the analysis window and the synthesis window being the same. If the window functions are the same, the aliasing components occurring during the sub-sampling can be compensated after the evaluation by the synthesis window.
To improve frequency resolution, it may be appropriate to employ a specially configured window for the analysis so as to realize, for example, a slight initial rise in the window function. The advantage of such a window function is that a very high frequency resolution is realized for narrowband signal components, leading to a very effective bit assignment with low data rate during coding.
The lecture manuscript by B. Feiten, "Spectral Properties of Audio Signals and Masking with Aspect to Bit Data Reduction", 86th AES Convention, March, 1989, discloses the utilization of different window functions for analysis and synthesis and to employ these for time sections that overlap by 50%. However, the described graphic definition of the synthesis function does not result in compensation of the aliasing components after evaluation by means of the synthesis window.